Leif Madsen [Tue, 2 Mar 2010 21:08:09 +0000 (21:08 +0000)]
Update IMAP documentation.
Update the IMAP documentation to make it clear that storing voicemails
in the same folder as a large number of emails could potentially cause
significant slow downs when writing or retrieving voicemails.
Jeff Peeler [Mon, 1 Mar 2010 17:02:03 +0000 (17:02 +0000)]
Modify queued frames from local channels to not set the other side to up
In this case, attended transfers were broken due to ast_feature_request_and_dial
detecting the channel being set to up before the answer frame could be read and
therefore failing to mark the channel as ready. This fix is a regression fix for
244785, which should continue to work properly as well.
Alec L Davis [Sat, 27 Feb 2010 23:51:28 +0000 (23:51 +0000)]
overlap receiving: automatically send CALL PROCEEDING when dialplan starts
Following Q.931 5.2.4
When the user has determined that sufficient call information has been received the
user shall stop T302 and send CALL PROCEEDING to the network.
Previously timeouts were possible if the dialplan took a long time to issue any
response back to the network.
Verified that our local TELCO also does the same.
(issue #16789)
Reported by: alecdavis
Patches:
based on overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Tilghman Lesher [Thu, 25 Feb 2010 21:21:05 +0000 (21:21 +0000)]
Some platforms clear /var/run at boot, which makes connecting a remote console... difficult.
Previously, we only created the default /var/run/asterisk directory at install
time. While we could create it in the init script, that would not work for
those who start asterisk manually from the command line. So the safest thing
to do is to create it as part of the Asterisk boot process. This also changes
the ownership of the directory, because the pid and ctl files are created after
we setuid/setgid.
(closes issue #16802)
Reported by: Brian
Patches:
20100224__issue16802.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir
Tilghman Lesher [Wed, 24 Feb 2010 21:02:18 +0000 (21:02 +0000)]
Remove color code sequences from verbose messages that go to logfiles.
(closes issue #16786)
Reported by: dodo
Patches:
logger2.patch uploaded by dodo (license 989)
Tested by: tilghman
Make chan_misdn DTMF processing consistent with other channel technologies.
The processing of DTMF tones on the receiving side of an ISDN channel is
inconsistent with the way it is handled in other channels, especially
DAHDI analog. This causes DTMF tones sent from an ISDN phone to be
doubled at the connected party.
We are using the following 2 options of misdn.conf
1) astdtmf=yes
2) senddtmf=yes
Option one is necessary because the asterisk DSP DTMF detection is better
than mISDN's internal DSP. Not as many false positives.
Option two is necessary to transmit DTMF tones end to end when mISDN
channels are connected to SIP channels with out of band DTMF for example.
The symptom is that DTMF tones sent by an ISDN phone are doubled on the
way through asterisk when two mISDN channels are connected with a Local
channel in between or if it is bridged to an analog channel.
The doubling of DTMF tones is because DTMF is passed inband to asterisk by
the mISDN channel and passed out of band once again after the release of
the DTMF tone. Passing it inband is wrong. Neither an analog channel nor
SIP channel passes DTMF inband if configured to inband DTMF. Analog and
SIP channels filter out the DTMF tones because they use the voice frames
returned by ast_dsp_process. But chan_misdn passes the unfiltered input
voice frames instead.
To overcome one aspect of the problem, the doubling of DTMF tones when two
mISDN channels are directly bridged, someone made an 'optimization', where
in that case the DTMF tone passed out-of-band to the peer channel is not
translated to an inband tone at the transmit side. This optimization is
bad because it does not work in general. For example, analog channels or
mISDN channels when bridged through an intermediary local channel will
generate DTMF tones from out-of-band information. Also, of course, it
must not be done when there is no inband DTMF available.
This patch fixes the issue. Now chan_misdn will filter the received
inband DTMF signal the same as other channel types.
Another change included: No need to build an extra translation path
because ast_process_dsp does it if required.
Russell Bryant [Thu, 18 Feb 2010 04:19:01 +0000 (04:19 +0000)]
Tweak argument handling for wget in the sounds Makefile.
1) Fix the check to see if we are using wget to not be full of fail. The
configure script populates this variable with the absolute path to wget if
it is found, so it didn't work.
2) Allow some extra arguments to be passed in for wget. This is just a simple
change to allow our Bamboo build script to tell wget to be quiet and not fill
up our logs with download status output.
David Vossel [Fri, 12 Feb 2010 23:30:17 +0000 (23:30 +0000)]
lock channel during datastore removal
On channel destruction the channel's datastores are removed and
destroyed. Since there are public API calls to find and remove
datastores on a channel, a lock should be held whenever datastores are
removed and destroyed. This resolves a crash caused by a race
condition in app_chanspy.c.
(closes issue #16678)
Reported by: tim_ringenbach
Patches:
datastore_destroy_race.diff uploaded by tim ringenbach (license 540)
Tested by: dvossel
David Vossel [Tue, 9 Feb 2010 22:55:38 +0000 (22:55 +0000)]
Fixes iaxs and iaxsl size off by one issue.
2^15 = 32768 which is the maximum allowed iax2 callnumber.
Creating the iaxs and iaxsl array of size 32768 means the maximum
callnumber is actually out of bounds. This causes a nasty crash.
Jeff Peeler [Thu, 4 Feb 2010 23:20:21 +0000 (23:20 +0000)]
Change channel state on local channels for busy,answer,ring.
Previously local channels channel state never changed. This became problematic
when the state of the other side of the local channel was lost, for example
during a masquerade. Changing the state of the local channel allows for the
scenario to be detected when the channel state is set to ringing, but the peer
isn't ringing. The specific problem scenario is described in 164201. Although
this was noted on one of the issues, here is the tested dialplan verified to
work:
Tilghman Lesher [Sun, 24 Jan 2010 06:33:01 +0000 (06:33 +0000)]
Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
Changed after discussion on the -dev list about possible unnecessary build
failures, due to checkouts/untars causing these special source files to
possibly be newer than their resulting C files. This should additionally
ensure that nobody need learn about extra Makefile arguments to ensure the
proper files get rebuilt when changes are made to these special source files.
Sean Bright [Wed, 20 Jan 2010 14:12:45 +0000 (14:12 +0000)]
Modify fix for issue 16554 to be more inline with what is already in trunk.
I should have taken a closer look at trunk/1.6.x, as this bug has already been
fixed in a much more simple manner, by just settings o->vars to NULL after the
ast_pbx_outgoing_* calls.
Sean Bright [Wed, 20 Jan 2010 14:06:49 +0000 (14:06 +0000)]
Fix a memory leak in pbx_spool when using SetVar in a call file.
In pbx_spool, when we are freeing our 'outgoing' struct, we weren't deallocating
the ast_variable list we had built from SetVars in a call file. Adding a call to
ast_variables_destroy in our deallocation routine works, but only if the variables
have not already been passed into ast_pbx_outgoing_app() or _exten(), both of
which take care of destroying the variable list for us.
Jeff Peeler [Tue, 19 Jan 2010 17:22:18 +0000 (17:22 +0000)]
Fix deadlock in agent_read by removing call to agent_logoff.
One must always lock the agents list lock before the agent private. agent_read
locks the private immediately, so locking the agents list lock is not an
option (which is what agent_logoff requires). Because agent_read already
has access to the agent private all that is necessary is to do the required
hanging up that agent_logoff performed.
Sean Bright [Mon, 18 Jan 2010 19:54:19 +0000 (19:54 +0000)]
Plug a memory leak when reading configs with their comments.
While reading through configuration files with the intent of returning their
full contents (comments specifically) we allocated some memory and then forgot
to free it. This doesn't fix 16554 but clears up a leak I had in the lab.
David Vossel [Mon, 18 Jan 2010 16:51:35 +0000 (16:51 +0000)]
updated transmit_silence option documentation in asterisk.conf
This patch updates the transmit_silence option to better document
why the option exists, and what it affects. Thanks to russell
for providing the verbage for this update.
Tilghman Lesher [Fri, 15 Jan 2010 20:52:27 +0000 (20:52 +0000)]
Disallow leaving more than maxmsg voicemails.
This is a possibility because our previous method assumed that no messages are
left in parallel, which is not a safe assumption. Due to the vmu structure
duplication, it was necessary to track in-process messages via a separate
structure. If at some point, we switch vmu to an ao2-reference-counted
structure, which would eliminate the prior noted duplication of structures,
then we could incorporate this new in-process structure directly into vmu.
(closes issue #16271)
Reported by: sohosys
Patches:
20100108__issue16271.diff.txt uploaded by tilghman (license 14)
20100108__issue16271__trunk.diff.txt uploaded by tilghman (license 14)
20100108__issue16271__1.6.0.diff.txt uploaded by tilghman (license 14)
Tested by: jsutton
Jeff Peeler [Wed, 13 Jan 2010 19:43:33 +0000 (19:43 +0000)]
Fix regression for timed out parked call returning to caller
This issue seems to have been exposed by the fix in 160390 whereby using a
masquerade prevented a crash. The new channel used in the masquerade was
not copying the macro information from the old channel.
David Vossel [Wed, 13 Jan 2010 17:16:12 +0000 (17:16 +0000)]
add silence gen to wait apps
asterisk.conf's 'transmit_silence' option existed before
this patch, but was limited to only generating silence
while recording and sending DTMF. Now enabling the
transmit_silence option generates silence during wait
times as well.
To achieve this, ast_safe_sleep has been modified to
generate silence anytime no other generators are present
and transmit_silence is enabled. Wait apps not using
ast_safe_sleep now generate silence when transmit_silence
is enabled as well.
Tilghman Lesher [Tue, 12 Jan 2010 03:18:36 +0000 (03:18 +0000)]
Portability and other fixes for the safe_asterisk script
(closes issue #16416)
Reported by: bklang
Patches:
safe_asterisk-compat-1.patch uploaded by bklang (license 919)
20100106__issue16416__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: bklang
Tilghman Lesher [Sat, 9 Jan 2010 00:57:58 +0000 (00:57 +0000)]
-1 is interpreted as an error, intead of the maximum mask.
(closes issue #16241)
Reported by: vnovy
Patches:
manager.c.patch uploaded by vnovy (license 922)
Tilghman Lesher [Mon, 4 Jan 2010 18:19:00 +0000 (18:19 +0000)]
Add a flag to disable the Background behavior, for AGI users.
This is in a section of code that relates to two other issues, namely
issue #14011 and issue #14940), one of which was the behavior of
Background when called with a context argument that matched the current
context. This fix broke FreePBX, however, in a post-Dial situation.
Needless to say, this is an extremely difficult collision of several
different issues. While the use of an exception flag is ugly, fixing all
of the issues linked is rather difficult (although if someone would like
to propose a better solution, we're happy to entertain that suggestion).
(closes issue #16434)
Reported by: rickead2000
Patches:
20091217__issue16434.diff.txt uploaded by tilghman (license 14)
20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: rickead2000
Tilghman Lesher [Wed, 30 Dec 2009 21:57:10 +0000 (21:57 +0000)]
Don't queue frames to channels that have no means to process them.
(closes issue #15609)
Reported by: aragon
Patches:
20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14)
Tested by: aragon
Sean Bright [Mon, 28 Dec 2009 15:12:08 +0000 (15:12 +0000)]
Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces.
There was conditional code (based on build platform) to optioinally wrap
PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions
of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add
a configure-time check for it.
Sean Bright [Mon, 28 Dec 2009 12:43:36 +0000 (12:43 +0000)]
Avoid a crash with large numbers of MeetMe conferences.
Similar to changes made to Queue(), when we have large numbers of conferences in
meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and
crash, so instead just use a single fixed buffer.
David Vossel [Tue, 22 Dec 2009 16:58:19 +0000 (16:58 +0000)]
fixes issue with p->method incorrectly set to ACK
It is possible for a second ACK to come in for a retransmitted message.
If an ack does not match an unacked message in our queue, restore the previous
p->method as this ACK is completely ignored.
Jeff Peeler [Mon, 21 Dec 2009 19:43:41 +0000 (19:43 +0000)]
Change Monitor to not assume file to write to does not contain pathing.
227944 changed the fname_base argument to always append the configured monitor
path. This change was necessary to properly compare files for uniqueness.
If a full path is given though, nothing needs to be appended and that is
handled correctly now.
Tilghman Lesher [Mon, 21 Dec 2009 16:45:03 +0000 (16:45 +0000)]
Send parking lot announcement to the channel which parked the call, not the park-ee.
(closes issue #16234)
Reported by: yeshuawatso
Patches:
20091210__issue16234.diff.txt uploaded by tilghman (license 14)
20091221__issue16234__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: yeshuawatso
projects / thirdparty / asterisk.git / log
